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This week we’ll talk about randomness and it’s role in my creative process, as I use a lot of randomized effects and sequences in nearly all of my tracks and I would consider it to be an important aspect of my sound. It is very hard to achieve true randomness, but we can approximate to a degree with various methods of processing in Ableton Live.

When I work with midi instruments, I like to use the random effect, sometimes combined with the velocity effect also set to random. With a melodic or bass pattern of midi notes, setting the choice, steps, and subtract/add/both options, as well as a carefully chosen randomness percentage (the percentage of notes going through the effect that are modified), you can add just the right amount of change to the pattern without removing the melodic structure which makes it work. You can combine this with an arpeggiator as well, usually before the random instance. I also save these patterns to get some replicability when it comes to mixing the track, or simply to save some cpu cycles. You can do this by setting up a second midi channel to record the output of your randomized channel and crank the tempo up to maximum to do this super quick, and just disable or delete the midi effects on the primary channel and use the recorded midi to trigger your instrument. I like this with bt and iZotope’s fantastic Stutter Edit as well, as it is triggered by midi notes, so having a random midi file can create some great textures. You can listen to exactly that effect on my track out of.

Another one of my favourite methods to create completely different audio loops is to use clip launch triggers within session view, the exact settings of which are in the image, and you can change the trigger length as you like. This is best with more than six similar drum loops – I prefer top or bongo/conga loops but it works with anything. Drop a whole bunch into one session view track, set the clip launch settings to all of the clips as pictured, and record. Again, you can crank the tempo to the max to do this. Then, in arrangement view, you’ve got a nice long recording of random percussion hits to select the best bits from- or just use it all.

The last technique I will cover with this post is random lfos (use whichever you prefer), and the powerful multimap pro. Both of these are Max4Live devices so you will need to have suite to use them. You could use the multimapper included in the old Max4Live essentials pack, but the pro plugin allows you to draw response graphs for each parameter, in addition to the min/max control of the old multimapper. My response graphs start to look a little ridiculous, but it sounds fun and that’s the important thing. This allows one lfo to control up to six parameters, although technically more as one rack knob counts as one parameter. The response graphs allow the control change with the lfo to be non-linear, something I really appreciate – as you can see here (image). This helps me create really interesting textures in my sound design process, and can create some really fun and unique sounds to use in tracks.

That should cover the basics of my usage of randomness in my track writing process, although I did skip over some things so I may come back to this in the future.

Last week we went over some bass programming and processing techniques, and I mentioned parallel processing. I also mentioned it in some of the posts on eqs and filters, so this week we’ll dive further into parallel processing and what you can do with it.

While it may sound super fancy and complicated, parallel processing is something you are probably already doing. It is balancing a processed signal with an unprocessed signal, to get a mix of the effect which isn’t overpowering or destructive. The Dry/Wet control on most effects does this very easily, although there are other ways of going about creating a parallel chain. You can use Ableton’s effect racks to create an entire effects chain – not just a single effect instance – and mix it with a dry chain to add slight distortion, colour, or reverb to your sound. I do this very often with the GlitchMachines plugins mentioned in an earlier post, sometimes stacking 2-4 chains of glitchy mangled silliness with a dry chain playing just slightly louder.

Parallel compression may be what comes to mind first when talking about parallel processing. I think the easiest way to learn this is to duplicate whichever track you would like to parallel compress, and putting an aggressive compressor on one channel. You can use a preset for this, or program your own. I usually go for -8 to -12dB of gain reduction (make sure to turn off makeup gain and match levels yourself). On its own, this will almost always sound absolutely terrible, but now – while playing both the uncompressed and compressed channels – you’ll turn the compressed channel level way down, I would suggest to the point where you can’t hear it, and bring it back up until it starts to add some weight and volume to the mix. I would suggest starting with this duplicated track method until you are confident with the process, and then move to either a parallel effects rack or simply the dry/wet knob which is present on most compressors, again being careful to level-match so you can properly judge the efficacy of the parallel compression.

Parallel compression can be really valuable for adding weight to basses and kicks, punch to mid-range elements, and overall to add volume or (with the right compressor model) a little colour as well. Be careful on bass elements, as it’s very easy to kill the low-end entirely with phase cancellation. I use this technique on my high-hats buss in many mixes, with just a little bit of AudioDamage’s RoughRider2 compressor mixed in. I like that particular compressor for this because it softens some of the harshness which can come from extremely high-frequency sounds, and it can soften the transients just a touch without killing the punch or dynamics. This one is easy to overdo, but as anything, when used right it’s perfect.

You can also set up an equalizer or filter in parallel effects chains, to boost or cut some of the audio signal, or to add a resonance peak without disrupting too much of the sound. This can also add some phasing artifacts, but above the bass range, it isn’t always super detrimental. Use your ears, and you should be able to tell what is good and bad. In the mid to high-frequency range, some phasing issues are often perceived as stereo width, which can really open up your mixes and bring elements out of a track.

You can also experiment with sidechaining or gating the affected signal, to add some depth to the sound design and create space for both the dry sound and the effect channel. This can have a similar effect as chokes within a drum rack, or something like a trance gate, depending on the audio elements and effects used. I will sometimes gate a signal so just the peak comes through, and then use a delay or reverb on the peak; adding that touch of space to the mix without the mud and fuzziness which can also occur.

I may come back to parallel chains in a later post, but that covers the basics for now. I haven’t decided what the next few posts will cover yet, but tune back in next Monday for more production tips.

This week I have a quick post on some bass programming and processing techniques. Hopefully, this will give you a little more confidence when sequencing/designing your bass sounds, and how to process them – whether you made the sound from scratch or used a loop. I’ve already covered some main components of bass processing in the compression, distortion, and filter/eq posts, so I will be referencing back to those in this post.

It is best practice to work with basses in mono, or at least collapse anything below 100Hz to mono so as to work with a mid/side signal chain. This prevents phase issues, as covered in previous posts and later on in this post.

Getting the “groove” with a bass, kick, and other percussive elements just right could the most important element in a dance music track. Of course, rules are meant to be broken, and sounding good is the most important. I like to start with some premade loops, to establish what I want my groove to sound like, and I’ll often design a sound to work with the loop sound and use a similar groove. One thing to note when programming the loop is that you can often avoid a lot of extra processing later on with a little extra thought into the synth patch and note timing. A short release keeps the notes tidy and helps to avoid a muddy low-end. Keeping bass hits off the kick drum will allow the kick to breathe more – this can mitigate the need for as much sidechain compression, or possibly allow you to forgo sidechaining altogether.

Whether you are using a synth or a loop, distortion is incredibly useful in making the sound seem bigger, louder, and punchier, all without adding to the volume at all. Because of how hearing works, and how tracks can be mixed, some harmonics, or distortion harmonics, in the 500-1500Hz range can be used to clue in the listener to the root bass sound occurring below, sometimes as low as 40Hz. This frees up headroom and allows for more mix clarity, as we have “boosted” the low end of the bass sound simply by adding some dirt in the higher frequency ranges. To get this sort of dirt, I’ll use frequency modulation, analogue filter models with a few dB of drive, a saturator or other distortion plugin, or layer another sound onto the bass.

Filters and equalizers are, of course, also important in sculpting the sound to both sound right and fit into the mix. Be careful about phase issues, as I mentioned in the filter/eq posts, as phase issues can be especially detrimental in the lower frequencies. It is also entirely possible your monitors or headphones cannot go low enough to properly represent lower frequencies as well, and the room you work in can also introduce phase issues based on the room size and geometry. Filtering the ultra-low sounds out of your bass can help with clarity though, anything below the lowest note played can be cut off, but again be aware you don’t lose anything to phase cancellation. Using some gentle eqs on the fundamental frequency, to add some weight, or cutting the top end a little is also powerful – I would suggest doing this after any distortion plugins and before any compressors. A resonant low-pass filter is also commonly used. The typical acid sound comes from a particular low-pass resonance filter circuit.

I will only cover distortion here briefly. Load up your favourite distortion plugin, and start with a little. Sometimes multiple plugin instances at low dry/wet can give a better effect than one instance at full. It depends on the sound, the plugin, and the rest of the mix – but of course, your ears can always tell when it is too much.

Compression is the last main point here, and again probably the least understood. Compression is used to decrease dynamic range, or volume, and can allow us to turn up quieter sounds more. In my experience, basses work best with gentle compression. A low ratio and slow attack work best to compress more transparently but also add some body and weight to the mix. Gain reduction usually isn’t too high here, and I’ll gain-match before and after to determine if the compressor does indeed make the bass sound better. Parallel compression can also be used here – it can sound really good with an analogue compressor model and can add a little extra grit to the sound when it’s overdriven (or clipped). Just make sure to gain-match again if you are going with the parallel compression, and especially an overdriven one. If you aren’t quite sure what I am on about with parallel compression, I’ll cover it in depth next week.

Over the last two posts I gave you a run through of filters and equalizers, and their basic uses and best practices. This week I’ll finish up this series by talking about some modulation options to be explored with filters and equalizers. Modulating these processing tools is incredibly valuable in adding dynamics, depth, personality, and interest into your tracks and mixes, and in some cases can make the difference between an ok sound and an awesome sound.

Within most synths, both hardware and digital, there is a filter envelope control which can be set to positive or negative time influence on the sound. When I am creating my own synth patches, I’ll often ignore the amplitude envelope entirely after setting the fastest attack possible, and a decay/sustain to suit the rhythm of the track. The additional motion within my own synth patches is often entirely contained within the filter envelope and sometimes some lfo routed into the filter as well. By adjusting the time parameter from negative to positive on the filter envelope, we can completely change the feeling of the sound from one note trigger to the next. Of course, not all synths include the negative amount parameter, but most do. We can also do this with pre-recorded loops, and live’s built-in auto filter. The envelope there can even be triggered by a sidechain input to create a completely different filter rhythm than you would otherwise get from a standard envelope. These frequency cutoff modulations can be combined with drawn or recorded resonance modulations to add emphasis to the moving cutoff frequency.

As I briefly mentioned above, lfos can also be used to great effect with filters, changing (usually) the cutoff, resonance, or envelope parameter of a filter. Within most soft-synths this is really easy to route together, and if all else fails the lfo plugins available via max4live are incredibly powerful for this sort of modulation. I like using lfos to adjust parameters, but also going into live’s clip modulation window and changing the parameter modulation in some awkward, unlinked loop length. This way I can add a crazy amount of resonance or a weird cutoff jump maybe every eleven bars, or three and third bars; something completely unrelated to the structure of the track and the lfo rate. I usually stick with quantised and synchronised lfo rates as well, so this off-beat modulation can sound particularly powerful, as most of the modulations are locked to the grid.

To finish up the post for this week, let’s also briefly go over parallel equalizing. I’ll hit parallel processing in further depth in a future post, but this is a quick intro in relation to filtering sounds. If you are finding a filter modulation to sound a little too powerful, but it’s still an important element in a sound, you can mix in the modulation as a part of the full sound, and let unprocessed sound through as well. Some filter plugins will give you a dry/wet control, but most will not. Live gives you a quick workaround using effect racks, and I use these a lot when I am processing tracks. Simply create two chains within the rack, one wet and one dry, and adjust volumes to suit the mix you want. The filtered chain can be combined with compression, delay, reverb, or anything else which compliments the sound and the mix of the track overall. Again, I will get to parallel processing in further depth later on, but this is a quick idea as to how to incorporate the technique with filters and equalizers.

Tune back in next Monday for some more production tips, and don’t forget to check out my previous posts for more tips, tricks, and extra knowledge you can annoy your friends with.

Last week I dove into filters, and this week I will continue on that theme with equalizers. Equalizers seem to be one of the things beginners get hung up on, and one of the processing tools most useful for mixing.

At it’s core, an eq is a combination of various types of filters. We started with set-band eqs, with a programmed curve, resonance, and frequency band, and the only possible adjustment was gain. Since then audio engineers have made much advancement, and the world of digital has allowed us even more capability. The most common eq most people are familiar with in a daw is the parametric eq, in which there are multiple bands which can be set to any filter type, frequency, resonance, and a selection of slopes. We can use these simply as filters, cutting out unwanted frequencies, but we can also boost content we want to emphasize or a combination of techniques.

A question many people have when they start to work with eqs, especially parametric digital equalizers, is what do I boost, what do I cut, and by how much? The answer is, of course, it depends. The first, and most important thing to keep in mind is that eqing should always happen in the mix. This is important to ensure all elements of the track work together and sound awesome together. You could have the best kick sound in the world, but if it doesn’t gel with the rest of the mix, it doesn’t matter. So, with that in mind, I will run through a few tips for using equalizers.

Start by cutting the frequencies you don’t want in the sound, although some people will say you should not do this but instead choose samples without extraneous frequency content you do not want. I will engage with that argument in a later post.

After cutting, you can start to gently attenuate or boost parts of the sound you want to diminish or emphasize. Depending on the sound and what your ears tell you about how the sound works in the mix, you can use shelf filters, notch, or a band filter, or any appropriate combination. When boosting, be aware of how much gain you are adding to the sound. It is entirely possible you will start to clip the sound by boosting the volume over 0dB, and that will sound bad. If you still need to boost by so much, use the output gain on the filter, or use a utility after, to match the input and output gain. This is important, not only to prevent clipping but also to allow your ears and brain to make the right mixing choice. A louder sound will almost always sound better than a quieter sound, thus matching input and output volume is crucial in making accurate decisions about how an effect sounds and if it is necessary.

When using both an equalizer and a compressor on a track, it is best practice to locate the compressor after the eq. This is to ensure you are only compressing the heard sound, and the compressor is not attenuating the gain based on a louder frequency band which the equalizer is removing.

I will conclude this post by mentioning dynamic equalizers. Examples of such are the free tdr nova, and the excellent waves f6. A dynamic equalizer combines the filters of a parametric eq with the active gain attentuation of a compressor. Depending on what problem you are trying to solve, or what type of sound you wish to achieve, dynamic eqs can be incredibly useful. It allows you to boost or attenuate a frequency band, and then apply a compressor on just that band. The compressors range can often be inverted as well, to boost a frequency but only when it crosses a set gain threshold. These compressors can also be sidechained, to allow you to move one sound out of the way of another in a very precise manner. I wouldn’t necessarily use these on every track or even each project, but they are a useful tool to have and have some knowledge of how to use.

Check back in next week for some tips on creative sound design effects using filters and equalizers.

In the next few blog posts, I will briefly explain and offer some use case examples for a selection of tools and techniques which I find to be rather misunderstood. These tools will also combine together nicely, so in a few weeks’ time, you’ll have a better understanding of where to go with some of your creative and mixing decisions.

This week we will begin with filters, and next week I’ll finish on that theme by running through equalisers.

Filters are incredibly simple at first, and equally, incredibly important. As with all effects, they can be used both creatively and technically; correct use will not only clean up your mixes but can also add much more depth and interest. Filters usually come in four primary types and a few other flavours which are combinations of the four basic ones. A low pass filter is used to remove higher frequency content, thus the name; the lows pass. Conversely, there is also the high pass, which cuts out low frequencies. The next two can be viewed as combinations already, a notch filter allows all frequencies to pass except for a specified band; the notch. A band pass is the opposite, and will only allow a specified frequency band through. Your standard filter will have a few basic settings, filter type, cutoff frequency, resonance, and slope. More controls can be added when the filter is more advanced or designed for different purposes.

How do I decide which filter to use and where? As with any effect, the first thing to ask is, what would I like to do? It seems completely obvious, but it is completely common for many people to throw on a filter or eq, and a compressor, onto every channel – either because they can or someone told them that was a good idea. Some tracks and styles may need all that, sure, but most don’t. So start by deciding what you want to do, and if a filter helps you get there, we can move on. If it’s a simple cut of offending or muddying frequencies, we can simply make that cut, but we don’t have to stop there. Filters have always been used as creative tools as well as precise technical tools. For my ears, this means using specific analogue-modelled filters to achieve a certain colour or distortion to the sound. Clean digital filters are fantastic for clean adjustments to sounds, but analogue-modelled plugins (or better yet, the real thing, if you can afford it) can add that last little bit to bring a sound out of the mix or give it a little something extra. The classic example of this is the classic Moog filter. Depending on the model and derivation of the particular plugins you have available, the two key parameters for such colouration are the resonance, that is the emphasis placed on the cutoff frequency, and the filter drive if available. The character of the sound will change with modifications to these parameters and will depend on the analogue model used. As an easy way to play around with the different modelling types, Live’s built-in auto filter device uses a few, which are also available in the filter sections of the operator, simpler, and sampler midi instruments. Ableton themselves are a little skittish about saying exactly which filters they modelled, which include the famous ms-20 and moog designs.

As a final note, I will briefly cover some of the inherent issues with filters. They cannot cut absolutely surgically, for example, a sound cannot be at 0dB at 400Hz, and cut to -∞dB at 410Hz. Filters need some space to cut, and that is called the slope. Slopes generally range from -6dB per octave to -48dB/oct. The steeper you go, the more phasing issues can occur, as the wave phase at the cut frequency range is affected by the filter. For this reason, cutting too steeply, with too many consecutive filters, or at very low frequencies can be very detrimental to sound quality. In most cases, it is not a huge issue, but something to be aware of.

Next week we will continue on this theme with how equalisers work and how to best use them.

Resampling, or recording processed audio back into the daw, is an important technique for my specific style of effect use and sound design. This can be done easily in a few ways within Ableton Live. Firstly, using a fresh audio track, setting the input to Resample, and arming the track to record. This will record any sounds coming from the master output when recording is initiated. Alternatively, you can also select a specific track within the project for the blank recording track to take as an input. Finally freezing and flattening is always an option, but this can limit your adjustment potential later on. If you do freeze and flatten, I recommend saving the specific audio track before freezing, so you have the option to adjust the original processing later if needed.

With resampled audio, the immediate benefit is to decrease computer resource usage on midi instruments and audio effects, freeing up your computer to be more responsive or to add further effects. With my resampled audio I will often warp some small segments in excessive ways and with technically inappropriate warping modes, to achieve a unique texture or glitchy effect. This of course rapidly degrades the audio quality, but in this case we want to degrade the quality to create previously unheard or unattainable sounds.

The last technique I will run over this week is reversing. This can be done simply to reverse the sound, or we can apply effects on the sound in reverse, resample, and re-reverse the sound. This gives us some completely strange artifacts and textures otherwise unachievable. It’s commonly used in horror films for example; try this with a nice long reverb and you will immediately hear the “creepyness.” I’ll also do this with delays at times, to build up into a transition or intoduce a sound in a unique way. Combine the “pre” delay with some repitched delay time modulation and you’ve got even more madness to explore.